1. Field of the Invention
The invention relates to a method and a system for transmitting at least one data stream of a multimedia session through a distributed telecommunications network comprising at least one network core comprising a plurality of nodes interconnected by a plurality of links, each node of the network having a routing module, the session being established between a first terminal and at least one second terminal outside the network core, and the network core also comprising at least two border modules, making up points of access to the network core for the first and second terminals.
2. Description of the Related Technology
In the rest of the description and in the claims, “terminal” refers to equipment capable of sending and/or receiving data, whether it involves a user terminal or security equipment, acting as a proxy for terminals deployed within a protected perimeter and only accessing networks outside the perimeter through that security equipment.
Embodiments in particular apply to the real-time transmission of multimedia data on a communication network using the IP protocol, the data transmission in the network core being done using radio links.
Such a network comprises variable throughput links, in particular due to the variation of the propagation channels, for example due to weather conditions, and the sharing of spectral resources, consisting of dynamically allocating the spatial resources as a function of instantaneous needs. The radio resources are thus limited, variable, and costly.
Furthermore, real-time multimedia data transmission is demanding in terms of quality of service. In fact, such data is sent in the form of isochronous streams, the data packets of which are sent by a transmitting terminal at regular time intervals. These packets must be able to be received by a receiving terminal at regular time intervals for them to be usable.
Real-time multimedia stream transmission also requires monitoring several parameters, such as the bandwidth, the transit time, and the jitter, i.e. the variation in transmission timeframes of the packets on the network. Monitoring of these parameters throughout the length of the session defines the quality of service (QoS), which for the receiving terminal is expressed by a quality of experience (QoE).
“All IP” networks, which make it possible to have all of the telecommunications services converge on a same infrastructure, are not natively capable of providing such quality of service, in particular because they are based on individual data packet processing mechanisms, and not streams of packets.
Several mechanisms are therefore generally used to monitor the quality of service of sessions on such networks. In particular, the available bandwidth may be monitored by implementing a connection admission control (CAC), which makes it possible to determine before the transmission of streams whether the bandwidth available on the network is sufficient to transmit those streams with the required quality of service, without affecting the quality of service of other flows transmitted on the network, and to reject flows that would be in excess.
However, the admission control does not always guarantee that the requested bandwidth will be maintained during transmission of the stream, in particular when that transmission is done on wireless networks, the links of such networks often having a variable throughput. Thus, admission control does not make it possible to guarantee the availability of the bandwidth for the transmission of an entire stream, and decreasing the available bandwidth of a variable throughput link, on which streams admitted on that link at the end of an admission control are transmitted, may cause a loss of packets from those streams.
In order to guarantee the available bandwidth for transmitting streams, the admission control may be coupled with a resource reservation protocol. For example, the RSVP (Resource reSerVation Protocol) is a protocol that allows the recipient of data streams to request a certain quality of service (for example, the timeframe or bandwidth) through the network. This signaling protocol makes it possible to allocate the bandwidth dynamically: it is used by “real-time” applications to reserve the necessary resources at the routers for the necessary bandwidth to be available during transmission.
Furthermore, session control mechanisms make it possible to initialize, modify, and end multimedia sessions. For example, the session initiation protocol (SIP) makes it possible to authenticate and locate the participants in the multimedia session, the characteristics of the session being described using the SDP (Session Description Protocol).
The protocols used to transmit data streams generally use a layer-based architecture, each layer being responsible for providing one or more specific services to the layer above it, and communications being done only between adjacent layers.
However, in wireless networks, this organization is not optimal, the properties of the various layers being dependent on one another. In order to meet the quality of service requirements for multimedia sessions, the communication system must be able to adapt dynamically to traffic situations and network conditions, this need not being able to be met by the traditional protocol network architecture.
The emergence of a new concept, called “cross-layering,” allowing the layer-based protocol architecture to be violated, makes it possible to improve the transmission performance in wireless networks and ensure better quality of service for multimedia sessions. Thus, antenna systems, spectral distributions, modulation and encoding functions, information routing along multiple paths, and end-to-end optimization functions all adapt continuously as a function of the application needs and available capacity of the network.
Furthermore, the stream routing, admission control, and session control mechanisms are generally implemented separately, and do not allow the rapid deployment of networks using adaptive transmission systems.
Known from document WO/2008/125437 is a method for routing data streams in a network comprising multi-topology routers, each of the topologies being associated with a particular metric (for example timeframe, available throughput, bandwidth, etc.), in which the routing and reservation mechanisms are coupled. According to this method, a variation in the usable throughput of a link of the network causes a change in a metric, that change being taken into account to route the data streams.
This method has several drawbacks. In particular, it does not make it possible to optimize the network resources, and the variability of the throughputs of the links may cause signaling and processing overloads for the routers of the network. Furthermore, knowledge of the status of the network is not used during the negotiation of a service for a session, and the routing policies cannot be selected on a session-by-session basis.